The present invention pertains to a method for reproducing true source sound from acoustic speakers using digital signal processing (DSP) technology.
With the unprecedented increase in the use of multimedia information and the introduction of high-definition digital TV broadcasting, high-quality acoustic sound has become an essential element for producing a more realistic feel over what was produced in the past.
Acoustic waves generated by conventional acoustic speakers contain significant distortion because of the physical limitations on the mechanical structure of the speaker. Regardless of careful processing in-studio, or recording high-quality digital data using media such as digital audio tapes (DAT) or compact disks (CD), actual acoustic waves generated by conventional acoustic speakers are normally very different from the original source sounds and from high-precision reproductions of raw source sounds.
FIG. 1a shows a typical conventional acoustic system, and FIG. 1b shows typical frequency characteristics. In these FIGS., source sounds recorded on sound source 100, such as a compact disk, are converted to acoustic output by acoustic speaker 104 via amplifier 101, control amplifier 102, and power amplifier 103. Distortion in the reproduced audio is produced in the various stages, that is, the amplifiers 101, 102, and 103, and acoustic speaker 104.
FIGS. 2a and b represent the difference between original source sound and acoustic output measured using a reference microphone.
FIG. 3 shows an acoustic reproduction system defined as a signal processing box provided with input/output characteristics H(xcfx89). This acoustic reproduction system can be viewed as unknown box 110 that has known (measurable) input/output characteristics (distortion).
So this creates the problem of whether an additional signal processing box or filter that has inverse characteristics Hxe2x88x921(xcfx89) can be placed somewhere in this system, and if the original high-quality source sound can be restored in the output from acoustic speakers.
A filter is used as the measure for solving the present problem. FIG. 4 shows an acoustic speaker system that uses an inverse filter for high-quality acoustic reproduction. As is clear from the FIG., filter 115 with inverse characteristics Hxe2x88x921(xcfx89) is inserted between power amplifier 103 and acoustic speaker 104.
In the past, some research was carried out to achieve an ideal system that has flat amplification and linear phase characteristics. In theory, there are two approaches to this. In short, there is a time domain algorithm and a frequency domain algorithm. All approaches up to now have used only time domain algorithms. With this approach, the inverse filter coefficient is found with the method shown in FIG. 5, and filter coefficient wIK is updated adaptively so that the difference between original input audio signal 120 and output 121 from an acoustic is minimized. Filter coefficient calculation circuit 122 in FIG. 5 illustrates calculation of filter coefficient wIK with least squares method (LMS) algorithm 123 and the coefficient found here is given to filtering processing circuit 124.
The Kalman filtering theory is well-known. With this, updating of filter coefficient wIK is orthogonal to filtering error ek. This error ek is the difference between output delayed by delay circuit 125 and output filtered by filtering processing circuit 126 of filter coefficient calculation circuit 122. Delay zxe2x88x92xcex94is required to compensate for filtering delay. Expressed mathematically, filter coefficient updating is given by the equation below.       w    ⁡          (              k        +        1            )        =            w      ⁡              (        k        )              +                  α                                            x              ⁡                              (                k                )                                      T                    ⁢                      x            ⁡                          (              k              )                                          ⁢              e        ⁡                  (          k          )                    ⁢              x        ⁡                  (          k          )                    
Here, filter coefficient wIK and input signal xK are both given in the form of specific vectors.
The filter coefficient can be found by using either actual audio signals or reference white noise as input. Use of the latter has been shown to give a more accurate filter coefficient. However, in a dynamic environment, acoustic speaker characteristics must be compensated for adaptively, and actual signals must be used as input and the filter coefficient must be calculated adaptively.
A conventional approach of this type that obtains filter coefficients in time domain is simple, so it is often applied to actual practice. However, it has the important problems described below.
FIG. 6 illustrates typical actual acoustic speaker characteristics. It shows equalizable frequency range and unequalizable frequency range.
Due to physical limitations of acoustic speakers, a general characteristic of acoustic speakers, as shown in FIG. 6, is that they undergo significant attenuation in very low frequency bands and in very high frequency bands. This phenomenon is caused by physical limitations in speaker structure, and since it is generally a nonlinear property, there is the concern that trying to restore these characteristics will lead to acute deterioration in all speaker characteristics.
A conventional approach to recover acoustic signals in time domain tries to extract filter coefficients independently from this phenomenon. For this reason, rather than being satisfactory, it has led to gradual unexpected deterioration in acoustic speaker characteristics.
The present invention, in order to solve the aforementioned problems in the prior art, uses an audio signal processing method where the entire frequency band of input audio signals is divided into multiple sub-bands, equalizable sub-bands are recognized from each sub-band and at the same time a filter coefficient is found by comparing the equalizable sub-bands and corresponding sub-bands from output audio signals, and frequency convolution calculates a filter coefficient for the calculated sub-band, and input audio signals are processed based on this convolution.
Also, the audio signal processing system associated with the present invention has a sub-band analysis filter bank that divides the entire frequency band of input audio signals into multiple sub-bands, a filter coefficient calculating circuit that identifies equalizable sub-bands from each sub-band and at the same time calculates filter coefficients by comparing equalizable sub-bands and corresponding sub-bands from the output audio signals, and a processing circuit that performs frequency convolution on calculated filter coefficients for equalizable sub-bands and processes input audio signals based on this convolution.
Also, the digital signal processing device associated with the present invention is equipped with a data memory for storing data, a program memory for storing command programs, a multiplier, and a control unit. The aforementioned control unit enables control of writing of data to the aforementioned data memory and control of the aforementioned multiplier in response to command programs stored in the aforementioned program memory. The aforementioned data memory stores data for multiple sub-bands obtained by dividing input audio signals into multiple frequency bands and data for multiple sub-bands obtained by dividing reference audio signals into multiple frequency bands. The aforementioned program memory stores command programs for multiplying the ratio of the data for each sub-band of the aforementioned audio signals and the data for each sub-band of the aforementioned reference audio signals by the aforementioned filter coefficient in order to correct filter coefficients found from each sub-band for the aforementioned input audio signals and from each sub-band for the aforementioned reference audio signals, and for storing the results of said multiplication. The aforementioned control unit enables multiplication of the aforementioned ratio and the aforementioned filter coefficient by the aforementioned multiplier and stores the corrected filter coefficient in the aforementioned data memory.
Here xe2x80x9cequalizexe2x80x9d means to restore distorted signals to original signals.
With the present invention, the entire frequency band of the input audio signal is divided into multiple sub-bands and only equalizable sub-bands from among those sub-bands are equalized to find a filter coefficient. So it will be possible to filter audio signals appropriately and to reproduce original source sound with high precision, without causing deterioration/attenuation in low frequency and high frequency bands of the acoustic speaker as in the past.